A new delayless subband adaptive filter structure

Adaptive subband techniques have been developed to reduce complexity and slow convergence problems of the traditional fullband high-order adaptive filters. Some of the disadvantages often encountered in most of the proposed architectures are the effect of aliasing associated with the multirate structure, which is a source of error in the modeling of the unknown system, and the delay introduced in the signal path. We present a new delayless maximally decimated structure where the optimal subband filters are related to the wideband system in a closed form. They make use of a special DFT analysis filterbank where the polyphase components of the prototype filter represent fractional delays so that there is no need for adaptive cross-filters, and the unknown system can modeled perfectly in a closed-loop scheme. We interpret the proposed structure as a special case of a block adaptive filter with lower computational complexity than the conventional fullband LMS algorithm. Some computer simulations are presented in order to verify the good features of the proposed structure     

Robust microphone arrays using subband adaptive filters

A new adaptive beamformer which combines the idea of subband processing and the generalised sidelobe canceller structure is presented. The proposed subband beamformer has a blocking matrix that uses coefficient-constrained subband adaptive filters to limit target cancellation within an allowable range of direction of arrival. Simulations comparing the fullband and subband adaptive beamformers show that the subband beamformer has better performance than the fullband beamformer when the target and/or interfering signals are coloured. In reverberant environments, the proposed subband beamformer also performs better than its fullband counterpart     

Alias-Free Subband Adaptive Filtering With Critical Sampling

To overcome the limitations of a conventional fullband adaptive filtering, various subband adaptive filtering (SAF) structures have been proposed. Properly designed, an SAF will converge faster at a lower computational cost than a fullband structure. However, its design should consider the following two facts: the interband aliasing introduced by the downsampling process degrades its performance, and the filter bank in the SAF introduces additional computational overhead and system delay. In this paper, to fully exploit the benefits of using an SAF, an almost alias-free SAF structure with critical sampling is proposed. The interband alising is removed from the subband signal by isolating the aliasing using a bandwidth-increased analysis filter. Computer simulations show that the proposed structure converges faster than both an equivalent fullband structure at lower computational complexity and recently proposed SAF structures for a colored input.     

基于子带SDL的宽带自适应波束形成

针对宽带波束形成通常需要大量阵元或延迟线所带来的硬件开销较大,波束形成效率相对较低的问题,提出一种基于子带阵元延迟线（SDL）的宽带自适应波束形成算法。该算法首先建立子带SDL模型,然后利用分析滤波器组将阵元接收的宽带信号分解为子带信号并进行相应的子带线性约束最小方差（LCMV）波束形成,最后通过综合滤波器组得到全带的波束形成。仿真结果表明,子带波束形成不仅具有比全带波束形成更高的效率,更好的频率不变性,更强的抗干扰能力及更快的收敛速度,而且可以大大降低硬件开销。      

A Low-Complexity Delayless Selective Subband Adaptive Filtering Algorithm

Adaptive filters of significant order, requiring high computational complexity, are necessary in many applications such as acoustic echo cancellation and wideband active noise control. Successful approaches to lessen the computational complexity of such filters are subband methods, and partial updating schemes where only a part of the filter is updated at each instant. To avoid the time delay introduced by the subband-splitting, delayless structures which reconstructs a fullband filter, producing delayless output, from the adaptive subband filters have been proposed. This paper proposes a delayless subband adaptive filter partial updating scheme, where the general idea is to only update the most misadjusted subband filter(s). Analysis in terms of mean square deviation is presented and shows that the fullband filter convergence speed is significantly increased, even for flat spectrum signals, as compared to traditional periodic subband filter update with the same computational complexity. Echo cancellation simulations with an artificial system to verify the analysis, using both flat spectrum signals and speech, is also presented, as well as offline calculations using signals from a real system.      

A new approach to subband adaptive filtering

Subband adaptive filtering has attracted much attention lately. In this paper, we propose a new structure and a new formulation for adapting the filter coefficients. This structure is based on polyphase decomposition of the filter to be adapted and is independent of the type of filter banks used in the subband decomposition. The new formulation yields improved convergence rate when the LMS algorithm is used for coefficient adaptation. As we increase the number of bands in the filter, the convergence rate increases and approaches the rate that can be obtained with a flat input spectrum. The computational complexity of the proposed scheme is nearly the same as that of the fullband approach. Simulation results are included to demonstrate the efficacy of the new approach     

一种子带分频段信道估计算法

针对当前高速率通信中信道阶数很长导致信道估计和均衡困难的问题,利用子带 滤波器组近似完全重构的特点,提出一种在子带内进行分频段信道估计、在全频带综合信道 参数的估计方法。该方法较全频带信道估计收敛速度快,收敛误差小,能很好适应恶劣的信 道情况。虽然总的计算量大于全频带信道估计,但由于采用并行计算,所以能大大减少运算 时间。仿真试验表明,在重构误差足够小的情况下,子带数目越多,收敛越快,收敛残差比 全频带信道估计小5 dB左右。     

基于多相滤波器组的非线性回声抵消

在免提通话系统和移动通信设备中，扬声器常常工作在较高的音量下，容易发生过载现象，从而产生明显的非线性声学回声，这在小微型扬声器中更加常见。常用的线性AEC(Acoustic Echo Cancellation)算法无法消除此类非线性回声，因此通话质量受到严重影响。非线性回声主要表现为额外的高频谐波分量，这些分量使得全带系统不再满足线性关系，而通常的AEC算法都是基于最小化全带误差推导而来，因此性能很容易受到非线性失真的影响。本文提出了一种基于多相滤波器组的子带AEC算法，把全带误差变成了各个子带的误差，因而把谐波失真成分变成了某些子带内的加性噪声，这使得谐波失真较小的那些子带依然能够正常收敛。通过仿真和实测实验，当出现非线性失真时，新方法的ERLE(Echo Return Loss Enhancement)明显高于经典的全带时域和频域方法，对于非线性失真明显的语音信号，ERLE提升约10 dB。      

Linear prediction of subband signals

The performance of linear prediction of fullband and subband signals is described in terms of the respective prediction gain. The subband prediction gain is characterized in terms of the fullband signal power spectral density and the frequency response of the subband filters. For Gaussian fullband signals, the asymptotic subband prediction gain can never be larger than the asymptotic fullband prediction gain. Simulation results compare fixed and adaptive fullband and subband prediction gains for Gaussian sources and speech. For speech, the subband prediction gain can exceed the fullband prediction gain     

Subband adaptive filtering with real-valued subband signals foracoustic echo cancellation

Adaptive filtering in subbands was originally proposed to overcome the limitations of conventional least-mean-square (LMS) algorithms. In general, subband adaptive filters offer computational savings, as well as faster convergence over the conventional LMS algorithm. However, improvements to current subband adaptive filters could be further enhanced by a more elegant choice of their design/structure. Classical subband adaptive filters employ DFT-based analysis and synthesis filter banks which results in subband signals that are complex-valued. The authors modify the structure of subband adaptive filters by using single-sideband (SSB) modulated analysis and synthesis filter banks, which result in subband signals that are real-valued. This simplifies the realisation of subband adaptive filters     

利用直接矩阵求逆和临界采样子带自适应滤波器的快速系统辨识

在许多应用中,子带自适应滤波器结构已经显示了其在计算和性能上的优点。基于最近提出的一个采用临界采样滤波器组的子带自适应结构,该文引入了子带直接矩阵求逆(DMI)算法。在保持了该算法快速收敛优点的同时,利用相关矩阵块三对角的特殊结构,降低了该算法的计算复杂度。理论分析及计算机实验显示,子带直接矩阵求逆算法只需经过较少的更新次数自适应子滤波器自由度的两倍,就能够收敛到高于最小均方误差的3dB附近。     

Analysis of linear prediction, coding, and spectral estimation fromsubbands

Linear prediction schemes make a prediction xˆ_i of a data sample x_i using p previous samples. It has been shown by Woods and O'Neil (1986) as well as Pearlman (1991) that as the order of prediction p→∞, there is no gain to be obtained by coding subband samples. This paper deals with the less well understood theory of finite-order prediction and optimal coding from subbands which are generated by ideal (brickwall) filtering of a stationary Gaussian source. We first prove that pth-order prediction from subbands is superior to pth-order prediction in the fullband, when p is finite. This fact adduces that optimal vector p-tuple coding in the subbands is shown to offer quantifiable gains over optimal fullband p-tuple coding, again when p is finite. The properties of subband spectra are analyzed using the spectral flatness measure. These results are used to prove that subband DPCM provides a coding gain over fullband DPCM, for finite orders of prediction. In addition, the proofs provide means of quantifying the subband advantages in linear prediction, optimal coding, and DPCM coding in the form of gain formulas. Subband decomposition of a source is shown to result in a whitening of the composite subband spectrum. This implies that, for any stationary source, a pth-order prediction error filter (PEF) can be found that is better than the pth-order PEF obtained by solving the Yule-Walker equations resulting from the fullband data. We demonstrate the existence of such a “super-optimal” PEF and provide algorithmic approaches to obtaining this PEF. The equivalence of linear prediction and AR spectral estimation is then exploited to show theoretically, and with simulations, that AR spectral estimation from subbands offers a gain over fullband AR spectral estimation     

New prediction schemes for lossless coding of fullband and subband images

This paper presents a study of lossless image compression of fullband and subband images using predictive coding. The performance of a number of different fixed and adaptive predictors are evaluated to establish the relative performance of different predictors at various resolutions and to give an indication of the achievable image resolution for given bit rates. In particular, the median adaptive predictor is compared with two new classes of predictors proposed in this paper. One is based on the weighted median filter, while the other uses context modelling to select the optimum from a set of predictors. A graphical tool is also proposed to analyse the prediction methods. Simulations of the different predictors for a variety of real world and medical images, evaluated both numerically and graphically, show the superiority of median based prediction over this proposed implementation of context model based prediction, for all resolutions. The effects of different subband decomposition techniques are also explored.     

Architectural Synthesis of Computational Engines for Subband Adaptive Filtering

Architectural synthesis of low-power computational engines (hardware accelerators) for a subband-based adaptive filtering algorithm is presented. The full-band least mean square (LMS) adaptive filtering algorithm, widely used in various applications, is confronted by two problems, viz., slow convergence when the input correlation matrix is ill-conditioned, and increased computational complexity for applications involving use of large adaptive filter orders. Both of these problems can be overcome by the use of a subband-based normalized LMS (NLMS) adaptive filtering algorithm. Since this algorithm is not amenable to pipelining, delayed coefficient adaptation in the NLMS updation is used, which provides the required delays for pipelining. However, the convergence speed of this subband-based delayed NLMS (DNLMS) algorithm degrades with increase in the adaptation delay. We first present a pipelined subband DNLMS adaptive filtering architecture with minimal adaptation delay for any given sampling rate. The architecture is synthesized by using a number of function preserving transformations on the signal flow graph (SFG) representation of the subband DNLMS algorithm. With the use of carry-save arithmetic, the pipelined architecture can support high sampling rates limited only by the delay of two full adders and a 2-to-1 multiplexer. We then extend this synthesis methodology to synthesize a pipelined subband DNLMS architecture whose power dissipation meets a specified budget. This low-power architecture exploits the parallelism in the subband DNLMS algorithm to meet the required computational throughput. The architecture exhibits a novel tradeoff between algorithmic performance (convergence speed) and power dissipation. Finally, we incorporate configurability for filter order, sample period, power reduction factor, number of subbands and decimation/interpolation factor in the low-power architecture, thus resulting in a low-power subband computational engine for adaptive filtering.     

Convergence analysis of alias-free subband adaptive filters based on a frequency domain technique

Convergence analysis of alias-free subband adaptive filters (SADFs) is presented based on a frequency-domain technique where instead of analyzing the adaptive algorithms in the time-domain, the averaging method and the ordinary differential equation (ODE) method are applied to the frequency-domain expressions of the adaptive algorithms converted by the discrete Fourier transform. As an alias-free SADF algorithm, the SADF proposed by Pradhan and Reddy is known. In this paper, this technique is first applied to the Pradhan's SADF. The stability of the Pradhan's SADF is verified in the frequency domain, and a simple formula to evaluate the mean square error (MSE) of the algorithm is theoretically derived. By using a slight modification, the technique can be applied to the two-band delayless subband adaptive filter (DLSADF) with the Hadamard transform. The stability condition and the MSE of the DLSADF with the Hadamard transform are also obtained. Simulation results of both algorithms show the validity of the theoretical results.     

A New Normalized Subband Adaptive Filter Algorithm with Individual Variable Step Sizes

Proposed is a novel variable step size normalized subband adaptive filter algorithm, which assigns an individual step size for each subband by minimizing the mean square of the noise-free a posterior subband error. Furthermore, a noniterative shrinkage method is used to recover the noise-free priori subband error from the noisy subband error signal. Simulation results using the colored input signals have demonstrated that the proposed algorithm not only has better tracking capability than the existing subband adaptive filter algorithms, but also exhibits lower steady-state error.     

基于子带粒子滤波的一种语音增强方法

结合多采样率系统理论中的子带技术与贝叶斯估计理论中的粒子滤波技术,提出了一种基于子带粒子滤波的语音增强方法。该方法首先将语音信号分解成子带信号,建立各子带信号的低阶时变AR模型;然后利用R-B粒子滤波估计时变AR模型参数,对子带信号进行滤波处理;最后根据滤波后的子带信号重构语音信号,实现语音增强。该方法通过子带分解降低了R-B粒子滤波中采样空间的维数,在降低计算量的同时,提高了语音增强系统的性能。计算机仿真结果验证了该方法的有效性。     

基于传声器阵列的近场宽带波束形成方法

提出了一种适合于近场宽带传声器阵列处理的空时子带波束形成新方法。该方法通过将一个空域子带阵列和时域子带多速率滤波器组合并来同时获得空域和时域两种子带系统的优点。计算机仿真比较了其与全带自适应波束形成方法的性能,结果表明该方法对干扰有较强的抑制能力,改善了宽带波束形成频域不一致问题,并且能够并行处理,为实际语音通信系统的应用提供了一种有利途径。     

Filter bank design for a subband adaptive filtering structure with critical sampling

Subband adaptive filtering structures are attractive in applications such as acoustic echo cancellation and channel equalization, due to their properties of decorrelating the input signal and reducing the computational complexity. Recently, a new adaptive filtering structure with critical sampling was proposed. In this paper, we describe an optimization procedure to select the analysis and synthesis filter banks of this new subband structure, so that minimum steady-state mean square error or fastest convergence rate can be achieved. Such filter-bank design method is based on a theoretical analysis of the convergence properties of the adaptation algorithm and uses a nonlinear optimization routine. Computer simulations illustrate the convergence improvements that can be obtained with the filter banks designed by the proposed method.     

An embedding approach to frequency-domain and subband adaptivefiltering

Frequency-domain and subband implementations improve the computational efficiency and the convergence rate of adaptive schemes. The well-known multidelay adaptive filter (MDF) belongs to this class of block adaptive structures and is a DFT-based algorithm. We develop adaptive structures that are based on the trigonometric transforms, discrete cosine transform (DCT) and discrete sine transform (DST), and on the discrete Hartley transform (DHT). As a result, these structures involve only real arithmetic and are attractive alternatives in cases where the traditional DFT-based scheme exhibits poor performance. The filters are derived by first presenting a derivation for the classical DFT based filter that allows us to pursue these extensions immediately. The approach used in this paper also provides further insights into subband adaptive filtering